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11
Genesys CTI Technical Discussion / record schedule(Genesys cloud)
« Last post by Gautham on April 11, 2025, 06:57:43 AM »
Dear all,

I am creating a callback campaign for missed calls. The recording is getting inserted to list successfully using trigger. As per requirement i have to callback the customer after 30 min from the time of record creation. I have created a column in list where i am inserting the time.

How i can create 30 min delay in dialing for each record? please help
12
I guess the user that is registered to your app needs permissions over the other apps. Read at least. Otherwise won't have access to them

Enviado de meu SM-S918B usando o Tapatalk

13
Third-party applications are not getting audit events from the Genesys Engage Config server.
what are the required permissions for that? As I noticed if I assigned a Super Administrator role to user that i am using in third party application to get the audit event, in that case, third-party application is getting the audit event from the Genesys Engage Config server.

But if i remove the Super Administrator role from the user and assign only read access in that case Third-party application is not getting audit events from the Genesys Engage Config server.

Please help us to identify what are the minimum required permissions so that the Third-party application will be able to get audit events from the Genesys Engage Config server.
14
***EDIT*** DB tablespace was full. DBAs solved the issue.

Hi all, hope everything's fine out there. Data Mart tables are not receiving records, currently. I can see chunks being regularly collected using DMA.

Apparently no evident errors on LOGs, processes and DB are up and running.


Which components should I aim to and which errors I should pay attention to, to troubleshoot this data flow?


Best,
Gef
15
Genesys CTI Technical Discussion / Softphone audio delay
« Last post by victor on March 27, 2025, 08:14:25 AM »
Hi, has anyone measured the audio delay introduced by Genesys softphone? We are measuring about 80-120ms per endpoint, which seems to be pretty high. We tried switching PCs but the audio delay seems to be pretty much within the same range. Has anyone else looked into it?
16
Check here for more info
https://r.tapatalk.com/shareLink/topic?url=http%3A%2F%2Fwww%2Esggu%2Ecom%2Fsmf%2Findex%2Ephp%3Ftopic%3D12017%2E0&share_tid=12017&share_fid=68945&share_type=t&link_source=app

Enviado de meu SM-S918B usando o Tapatalk

17
Hi Experts,

I have been working on gathering voice quality metrics for each call made via Genesys SIP Endpoint. Although there is a documentation to support we can use publish/collector model via setting in SipEndpoint config file.

Producing RTCP Extended Reports:
https://docs.genesys.com/Documentation/SESDK/9.0.0NET/Developer/ConfiguringforNET

But the part which i am not able to understand is collector. Is it a Trunk DN or a different application altogether? Can someone please suggest?

Thanks
Aayush
18
Ok, after some troubleshooting on SIP sessions it turns out that an ALG software was blocking randomly OKs coming from the SIP Server, preventing the ACK from the Asterisk endpoint and the actual connection between caller and agent parties.
19
Hi all, hope everything's fine out there.

Basically there's no REINVITE to join caller party and agent party, and the call stays up, mute. I'm trying to figure out why calls coming from an Asterisk through a SIP Trunk, randomly get this error:

13:17:18.932: SipDialog: event CONNECTED_TIMEOUT_ST, t=324581940, s=7, r=5, m=e6dede8 port=[PORT]
13:17:18.932: CID:CUUID>6647aa0e-ffeb-4b35-ad1b-844073c413af:01L0V41B74A87400R5F2E2LAES06KDI4:
13:17:18.932 SIPCONN([CALLER NUMBER]): HandleSipDialogEvent(CONNECTED_TIMEOUT_ST)
13:17:18.932 SIPCONN([CALLER NUMBER]): new transaction
13:17:18.932 SIPCONN([CALLER NUMBER]): Park(1,40)
13:17:18.932 SIPCONN([CALLER NUMBER]): Park::Error
13:17:18.932 SIPCONN([CALLER NUMBER]): SIPCONN(e6ded10,01L0V49B74A87400R5F2E2LAES0HISG1) -Tr(262198752,SipTransactionAcceptCall)
13:17:18.932: Tr(262198752,SipTransactionAcceptCall):failed
13:17:18.932: Sc(262198746):step 0, Tr(262198752,SipTransactionAcceptCall) - failed
13:17:18.932: Tr(262198746,SipScenario):failed
13:17:18.932: SIPCM: transaction Tr(262198746,SipScenario) failed
13:17:18.932: GetRegistration::Unable to resolve number for DN:dummy
13:17:18.932: PI: 00 S[IN]D[[CALLER NUMBER]]C[*D[[CALLER NUMBER]]]P[MSML]
13:17:18.933: SIPPARTY(10051): handle postponed media service operation
13:17:18.933: media service handling postponed
13:17:18.933: previous scenario Tr(262198746,SipScenario) is not cleaned up
13:17:18.933: Sc(262198746):step 0, Tr(262198851,SipTransactionDetachMediaService) - begin
13:17:18.933 SIPCONN(MSML): DetachMediaPeer
13:17:18.933 SIPCONN([CALLER NUMBER]): ClrMediaPeer
13:17:18.933 SIPCONN(MSML): set monitor deaef90, 0
13:17:18.933 SIPCONN(MSML): state e:1,p:3,s:0,c:0,rc:0,m:0
13:17:18.933: SipDialog: ClearCall(phone=0,state=7)
13:17:18.933: SipDialog::Terminate(state=7,reason=0)
13:17:18.933: SIPDLG[18771797]: register TRN[324589627]
13:17:18.933: Sending  [0,UDP] 508 bytes to [SERVERNAME]:[PORT] >>>>>
BYE sip:Genesys@[MEDIA SERVER IP]:[PORT] SIP/2.0
From: <sip:[CALLER NUMBER]@[SIP SERVER IP]:[PORT]>;tag=0061AB84-2B39-1483-9000-D95E270AAA77-47872976
To: <sip:MSML@[SIP SERVER IP]:[PORT]>;tag=6202EB9B-0000-D961-9341-07A0917C9258
Call-ID: 0061AB66-2B39-1483-9000-D95E270AAA77-40936834@[SIP SERVER IP]
CSeq: 3 BYE
Content-Length: 0
Via: SIP/2.0/UDP [SIP SERVER IP]:[PORT];branch=z9hG4bK0061AB8E-2B39-1483-9000-D95E270AAA77-128741076
Max-Forwards: 69
Route: <sip:0x1eb43b00@[SERVER IP]:[PORT];lr;[HOSTNAME]=1;idtag=010A5724>


Any clue?


Best,
Gef
20
If somebody is still interested in this topic, here is a draft C# implementation of a KVC conversion, based on post by Rutger.
Types handling have to be improved. Right now it works for strings, sub-kvc and ints. For other types I don't have examples on my system right now.

[code=csharp]
        private Dictionary<string, object> ReadKeyValueCollection(byte[] buffer, int offset = 0)
        {
            var kvc = new Dictionary<string, object>();
            int fp = offset;

            int length = BinaryPrimitives.ReverseEndianness(BitConverter.ToInt16(buffer, fp));
            fp += 2;
           
            int pairs = BinaryPrimitives.ReverseEndianness(BitConverter.ToInt16(buffer, fp));
            fp += 2;

            while (fp - offset < length)
            {
                int type = BitConverter.ToChar(buffer, fp);
                fp += 1;

                int keyLength = BinaryPrimitives.ReverseEndianness(BitConverter.ToInt16(buffer, fp));
                fp += 2;

                var key = ASCIIEncoding.Default.GetString(buffer, fp, keyLength);
                fp += keyLength;

                int valueLength = BinaryPrimitives.ReverseEndianness(BitConverter.ToInt16(buffer, fp));
                fp += 2;

                kvc[key] = ReadValue(type, buffer, fp, valueLength);
                fp += valueLength;
            }

            return kvc;
        }

        private object ReadValue(int type, byte[] buffer, int offset, int length)
        {
            // 0 - str, 1 - int, 2 - utf16???, 3 - lst
            switch (type)
            {
                case 1: return ReadInt(buffer, offset, length);
                case 3: return ReadKeyValueCollection(buffer, offset);
                default: return ASCIIEncoding.Default.GetString(buffer, offset, length);
            }
        }

        private object ReadInt(byte[] buffer, int offset, int length)
        {
            switch (length)
            {
                case 1: return BitConverter.ToChar(buffer, offset);
                case 2: return BinaryPrimitives.ReverseEndianness(BitConverter.ToInt16(buffer, offset));
                case 4: return BinaryPrimitives.ReverseEndianness(BitConverter.ToInt32(buffer, offset));
                case 8: return BinaryPrimitives.ReverseEndianness(BitConverter.ToInt64(buffer, offset));
                default: return BinaryPrimitives.ReverseEndianness(BitConverter.ToInt16(buffer, offset));
            }
        }
[/code]
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